Distributed processing of audio signals

ABSTRACT

Remote audio information corresponding to remote audio signals are received by a controller from a remote node. A level of transmit attenuation and a level of receive attenuation are generated by the controller using the remote audio information. The local audio signals are attenuated with a transmit attenuator at the level of transmit attenuation for transmission to a remote node. Remote audio signals received from the remote node are attenuated with a receive attenuator at the level of receive attenuation for playback at a local speaker.

INCORPORATION BY REFERENCE

This application is a continuation-in-part of U.S. patent applicationSer. No. 08/342,270, filed Nov. 16, 1994, which is acontinuation-in-part of U.S. patent application Ser. No. 08/340,172,filed Nov. 15, 1994, which is a continuation-in-part of U.S. patentapplication Ser. No. 08/157,694, filed Nov. 24, 1993, all of which areincorporated herein in their entireties by reference.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to audio conferencing, and, in particular,to computer-implemented processes and computer systems for processingaudio signals for open audio systems.

2. Description of the Related Art

A traditional telephone handset places the transmitter close to theuser's lips and couples the receiver tightly to the user's ear. Aspeakerphone or other so-called open audio systems replace the handsetwith a separate microphone and loudspeaker that can be set on a table afew feet from the user. One problem with speakerphones results fromacoustic coupling, where the sounds produced by the loudspeaker arepicked up by the local microphone and retransmitted back to the sender.The result can be an undesirable echoing in which a talker hears his ownvoice delayed by the transmission to and retransmission from a remotespeakerphone.

One known solution to the problems of acoustic coupling and echoing isvoice switching, in which only one direction of transmission is fullyactive at a time. Systems for achieving this one-way-at-a-time (i.e.,half duplex) communication determine which direction is to be active ata given time.

Referring now to FIG. 1, there is shown a circuit block diagram of aconventional speakerphone 10. FIG. 1 is based on FIG. 1 from the seminalarticle "Fundamental Considerations in the Design of a Voice-SwitchedSpeakerphone" by A. Busala (The Bell System Technical Journal, Vol.XXXIX, No. 2, pp. 265-294, March 1960). As shown in FIG. 1, the voltagelevels of local audio signals generated by microphone 12 are attenuatedby transmit attenuator 14 for transmission to a remote node. Similar,the voltage levels of remote audio signals received from the remote nodeare attenuated by receive attenuator 18 for play at loudspeaker 16. Theamount of attenuation applied by attenuators 14 and 18 is dynamicallycontrolled by control circuit 19.

To achieve the one-way talk state in which the user of speakerphone 10talks to the remote participant, control circuit 19 causes receiveattenuator 18 to attenuate significantly the remote audio signalsreceived from the remote node, while transmit attenuator 14 permitstransmission of the local audio signals to the remote node withoutsignificant attenuation. Similarly, to achieve the one-way listen statein which the user of speakerphone 10 listens to the remote participant,control circuit 19 causes transmit attenuator 14 to attenuatesignificantly the local audio signals, while receive attenuator 18permits playback of the remote audio signals without significantattenuation.

In conventional systems such as speakerphone 10, the levels ofattenuation are determined based on the volumes of the participants, asindicated by the voltage (or energy) levels of the remote and localaudio signals. Thus, when one conversation participant starts to speaklouder than the other, control circuit 19 controls the attenuationlevels of the transmit and receive attenuators 14 and 18 to switch fromthe talk state to the listen state, or vice versa.

One of the problems with conventional speakerphone systems relates tocutoff, where the system incorrectly or prematurely changes thedirection of communication, thereby cutting a talker off before he hascompleted. It is desirable therefore to provide voice-switchedspeakerphones that avoid or at least reduce undesirable cutoffs.

It is accordingly an object of the present invention to address theproblems associated with undesirable cutoffs when using speakerphones orother open audio systems that rely on voice switching to avoid acousticcoupling.

It is a particular object of the present invention to provide open audioprocessing for computer-based audio/video conferencing systems.

Further objects and advantages of this invention will become apparentfrom the detailed description of a preferred embodiment which follows.

SUMMARY OF THE INVENTION

The present invention is an apparatus for processing audio signals.According to a preferred embodiment, the apparatus comprises (a) atransmit attenuator for attenuating local audio signals generated by alocal microphone for transmission to a remote node; (b) a receiveattenuator for attenuating remote audio signals received from the remotenode for playback at a local speaker; and (c) a controller, electricallyconnected to the transmit attenuator and the receive attenuator, forcontrolling a level of transmit attenuation for the transmit attenuatorand a level of receive attenuation for the receive attenuator. Thecontroller receives, from the remote node, remote audio informationcorresponding to the remote audio signals. The controller uses theremote audio information to generate the level of transmit attenuationand the level of receive attenuation.

The present invention is also a computer-implemented process forprocessing audio signals. According to a preferred embodiment, remoteaudio information corresponding to remote audio signals are received bya controller from a remote node. A level of transmit attenuation and alevel of receive attenuation are generated by the controller using theremote audio information. The local audio signals are attenuated with atransmit attenuator at the level of transmit attenuation fortransmission to a remote node. Remote audio signals received from theremote node are attenuated with a receive attenuator at the level ofreceive attenuation for playback at a local speaker.

BRIEF DESCRIPTION OF THE DRAWINGS

Other objects, features, and advantages of the present invention willbecome more fully apparent from the following detailed description ofthe preferred embodiment, the appended claims, and the accompanyingdrawings in which:

FIG. 1 is a circuit block diagram of a conventional speakerphone;

FIG. 2 is a block diagram representing real-time point-to-pointaudio/video conferencing between two computer-based conferencingsystems, according to an embodiment of the present invention;

FIG. 3 is a circuit block diagram of the audio systems of the local andremote nodes of the conferencing network of FIG. 2;

FIG. 4 is a process flow diagram of processing implemented by thehistogram generators of FIG. 3;

FIG. 5 is a process flow diagram of processing implemented by the statemachine of FIG. 3 for each pair of frames of local and remote audiosignals to control the attenuation levels of the transmit and receiveattenuators of FIG. 3, according to one embodiment of the presentinvention;

FIG. 6 is a state diagram for the state machine of FIG. 3 for theprocessing of FIG. 5; and

FIG. 7 is a diagram of a discrete state machine, according to analternative embodiment of the state machine of FIG. 3.

DESCRIPTION OF THE PREFERRED EMBODIMENT(S)

Referring now to FIG. 2, there is shown a block diagram representingreal-time point-to-point audio/video conferencing between twocomputer-based conferencing nodes, according to one embodiment of thepresent invention. Each PC node has a conferencing system 100, a camera102, a microphone 104, a monitor 106, and a speaker 108. Theconferencing nodes communicate via a digital network 110. Eachconferencing system 100 receives, digitizes, and compresses the analogvideo signals generated by camera 102 and the analog audio signalsgenerated by microphone 104. The compressed digital video and audiosignals are transmitted to the other conferencing node via network 110,where they are decompressed and converted for play on monitor 106 andspeaker 108, respectively.

Conferencing system 100 may be any suitable conferencing system and ispreferably an Intel® microprocessor-based computer system configuredwith Intel® ProShare™ conferencing software and hardware. Microphone 104may be any suitable microphone for generating analog audio signals.Speaker 108 may be any suitable device for playing analog audio signals.

Digital network 110 may be any suitable network for transmitting digitalsignals between the conferencing nodes. In one embodiment of the presentinvention, digital network 110 is an ISDN network. In other embodiments,digital network 110 may be other types of networks, such as a local orwide area network.

Referring now to FIG. 3, there is shown a circuit block diagram of theaudio systems of the local and remote nodes of the conferencing networkof FIG. 2. FIG. 3 represents conferencing in which the local node 302 isconfigured with an open audio system (e.g., a speakerphone) while theremote node 304 is configured with a closed audio system (e.g., atelephone handset or headset). In an alternative embodiment of thepresent invention, both local and remote nodes may be configured withopen audio systems. The audio processing of FIG. 3 is distributed inthat there is an instantiation of the audio processing on each of theconference nodes connected by the digital network 110 and the audioprocessing on one node is interdependent of the audio processing on theother node.

Local audio signals generated by microphone 306 of local node 302 areattenuated by transmit attenuator 308 for transmission over network 110to remote node 304, where they are played back at speaker 310. Remoteaudio signals, generated by microphone 312 of remote node 304 andreceived by local node 302 over network 110, are attenuated by receiveattenuator 314 for playback at speaker 316. State machine 318 controlsthe level of attenuation (α) applied by transmit attenuator 308 to thelocal audio stream and the level of attenuation (1-α) applied by receiveattenuator 314 to the remote audio stream.

State machine 318 determines these attenuation levels based on theenergy levels of the local and remote audio streams (which indicatelevels of audio volume) and on histogram information for the local andremote audio streams. In one embodiment of the present invention, theenergy levels are based on the mean sum of the squares of the digitizedaudio signals within a particular time period (i.e., audio frame). Thehistogram information is used to characterize the energy levels of thebackground noise of the local and remote audio streams, which in turnare used to determine when the conference participants are talking. Thehistogram information for the local audio stream is generated byhistogram generator 320 of local node 302. The histogram information forthe remote audio stream is generated by histogram generator 322 ofremote node 304 and received by state machine 318 over network 110.

Referring now to FIG. 4, there is shown a process flow diagram ofprocessing implemented by histogram generator 320 of FIG. 3 for thelocal audio stream. Histogram generator 320 uses the local audio signalsthat are generated by microphone 306 over a specified period of time(e.g., 5 seconds) to generate a local histogram (step 402 of FIG. 4). Inone embodiment of the present invention, the histogram is logarithmbased where each histogram bin corresponds to a specified range ofdecibels (dB). Histogram generator 320 divides the local audio signalsfor the previous time period into a plurality of audio frames (e.g., oneaudio frame=20 msec of audio signals). The histogram is generated byfilling the histogram bins based on the energies of the individual audioframes.

After generating the local histogram for the specified time period,histogram generator 320 generates the variance of the local histogram(step 404). The variance is used to determine whether the local audiosignals for the previous time period correspond to background noisealone or background noise plus talking. In the paradigm for audioconferencing, background noise is assumed to be relatively uniform,white noise. In this paradigm, when a conference participant is nottalking, the histogram variance is relatively small. On the other hand,the histogram variance is relatively large when a conference participantis talking.

Therefore, if the local histogram variance for the previous time periodis less than a specified variance threshold (e.g., 3 dB) (step 406),then the local audio signals are assumed to correspond to backgroundnoise alone. In this case, the mean of the local histogram is generated(step 408) and used to establish a local background noise threshold(step 410). In one embodiment, the local background noise threshold is 5dB greater than the local histogram mean. This local background noisethreshold is retained and used by state machine 318 to control thetransmit and receive attenuators 308 and 314 (as described below).

If, on the other hand, the local histogram variance is not less than thevariance threshold (step 406), then the local audio signals for theprevious time period are assumed to correspond to background noise plustalking. In this case, the local background noise threshold is notupdated and processing returns to step 402. Histogram generator 320continues to generate and analyze histograms for subsequent time periodsto determine whether to update the local background noise threshold.

In the embodiment of FIG. 3, histogram generator 322 of remote node 304uses processing analogous to that of FIG. 4 to generate a remotehistogram for the remote audio signals that are generated by microphone312 and a remote background noise threshold indicative of the backgroundnoise level at remote node 304. In the embodiment of FIG. 3 in whichremote node 304 is a closed audio system, the remote background noisethreshold is preferably 20 dB greater than the mean of a remotehistogram that corresponds to background noise alone. In an alternativeembodiment in which both the local and remote nodes are open audiosystems, the remote background noise threshold is preferably 5 dBgreater than the mean of the remote histogram. The thresholds for openand closed audio systems are selected differently because, in the openaudio system, the talker is typically further away from the microphonethan in the closed audio system. Histogram generator 322 transmits theremote background noise threshold to state machine 318 over network 110.

In an alternative embodiment of the present invention, the remote nodehas no histogram generator. In this embodiment, the local node generatesits own histogram for the remote audio signals received over the network110 and does its own histogram analysis analogous to the processing ofFIG. 4 to generate the remote background noise threshold for the remoteaudio signals.

Referring now to FIG. 5, there is shown a process flow diagram ofprocessing implemented by state machine 318 of FIG. 3 for each pair ofaudio frames (consisting of one frame of local audio signals and oneframe of remote audio signals) to control the attenuation levels oftransmit and receive attenuators 308 and 314, according to oneembodiment of the present invention. In this embodiment, state machine318 maintains a single continuous state variable i which is used to setthe transmit and receive attenuation levels. The state variable i canassume any value between -j and j, where j is a specified constant(e.g., 8).

Referring now to FIG. 6, there is shown a state diagram for statemachine 318 of FIG. 3 for the processing of FIG. 5. When the statevariable i is between -j and -k, where k is a specified constant lessthan j (e.g., 2), state machine 318 is in the listen state 602 and theattenuators are configured to permit the local user to listen to theremote user talk. When the state variable i is between k and j, statemachine 318 is in the talk state 606 and the attenuators are configuredto permit the local user to talk to the remote user. When the statevariable i is between -k and k, state machine 318 is in the idle state604. The idle state is a transition state between the listen and talkstates in which the attenuators are configured to allow both local andremote users to talk.

State machine 318 analyzes the energy levels of the attenuated localaudio signals T and the remote audio signals R (as shown in FIG. 3) forthe most recent local and remote audio frames to determine the currentmode of audio conferencing. There are five different audio conferencingnodes, defined as follows:

No Talking: (T≦τ_(L)) & (R≦τ_(R))

Single Talk (Local): (T>τ_(L)) & (R_(L) 23 τ_(R))

Single Talk (Remote): (T≦τ_(L)) & (R≦τ_(R))

Double Talk (Local Wins): (T>τ_(L)) & (R>τ_(R)) & (T>R+D1)

Double Talk (Remote Wins): (T>τ_(L)) & (R>τ_(R)) & (R>T+D2)

where τ_(L) is the local background noise threshold, τ_(R) is the remotebackground noise threshold, and D1 and D2 are threshold parameters(e.g., 10 dB for both D1 and D2).

Referring again to FIG. 5, if the current audio conferencing mode is theno talking mode (step 502), then state machine 318 slowly moves thestate variable i towards the idle state (step 504). To move slowlytowards the idle state, if the state variable i is greater than 0, thenstate machine 318 decrements the state variable i by 1, and, if thestate variable i is less than 0, then state machine 318 increments thestate variable i by 1. If the state variable i is equal to zero, then itis left unchanged.

If the current audio conferencing mode is the single talk (local) mode(step 506), then state machine 318 quickly moves the state variable itowards the talk state (step 508). To move quickly towards the talkstate, state machine 318 increments the state variable i by l, where lis a specified constant greater than k (e.g., l=10) and where the statevariable i cannot be greater than j.

If the current audio conferencing mode is the single talk (remote) mode(step 510), then state machine 318 quickly moves the state variable itowards the listen state (step 512). To move quickly towards the listenstate, state machine 318 decrements the state variable i by l, where thestate variable i cannot be less than -j.

If the current audio conferencing mode is the double talk (local wins)mode (step 514), then state machine 318 quickly moves the state variablei towards the talk state (step 516). If the current audio conferencingmode is the double talk (remote wins) mode (step 518), then statemachine 318 quickly moves the state variable i towards the listen state(step 520). Otherwise, state machine 318 leaves the state variable iunchanged (step 522).

After determining the current audio conferencing mode and adjusting thestate variable i accordingly, state machine 318 sets the transmit andreceive attenuation levels based on the state variable i (step 524). Inone embodiment of the present invention, the sum of the transmit andreceive attenuation levels always correspond to unity gain. In thisembodiment, the attenuation level for transmit attenuator 308 of FIG. 3may be represented by α and the attenuation level for receive attenuator314 may be represented by (1-α).

As shown in FIG. 6, when the state variable i is less than ok, thenstate machine 318 is in the listen state 602 and state machine 318 setsα equal to 0. As such, the attenuation level of transmit attenuator 308is 0 and the attenuation level of receive attenuator 314 is 1. In thisconfiguration, the local audio signals generated by microphone 306 arefully attenuated and the remote audio signals are unattenuated to permitthem to be played back at speaker 316, thereby achieving one-way (i.e.,half duplex) communication.

When the state variable i is greater than k, then state machine 318 isin the talk state 606 and state machine 318 sets α equal to 1. As such,the attenuation level of transmit attenuator 308 is 1 and theattenuation level of receive attenuator 314 is 0. In this configuration,the local audio signals are unattenuated and the remote audio signalsare fully attenuated to prevent them from being played back at speaker316, thereby achieving one-way communication.

When the state variable i is between -k and k, then state machine 318 isin the idle state 604 and state machine 318 sets α as shown in thefollowing Equation (1):

    α=(i+k)/2k.                                          (1)

In this configuration, both the local and remote audio signals arepartially attenuated. The idle state provides a continuous lineartransition for α between the listen and talk states, where α varies from0 (for i=-k) to 1 (for i=k).

Referring now to FIG. 7, there is shown a diagram of a discrete statemachine 700, according to an alternative embodiment of state machine 318of FIG. 3. Discrete state machine 700 comprises the following fivestates defined by the present five sets of conditions:

(1) No Talking state 702: (T≦τ_(L)) & (R≦τ_(R))

(2) Single Talk (Local) state 704: (T>τ_(L)) & (R≦τ_(R))

(3) Single Talk (Remote) state 706: (T≦τ_(L)) & (R>τ_(R))

(4) Double Talk (Local Wins) state 708: (T>τ_(L)) & (R>τ_(R)) &[(T-τ_(L))-(R-τ_(R))]>D1)

(5) Double Talk (Remote Wins) state 710: (T>τ_(L)) & (R>τ_(R)) &[(R-τ_(L))-(L-τ_(R))]>D2)

where T is the energy of the current frame of attenuated local audiosignals, R is the energy of the current frame of remote audio signals,τ_(L) is the local background noise threshold, τ_(R) is the remotebackground noise threshold, and D1 and D2 are threshold parameters(e.g., 10 dB for both D1 and D2).

As shown in FIG. 7, the possible transitions between states in discretestate machine 700 are limited to the following:

From No Talking state 702 to either Single Talk (Local) state 704 orSingle Talk (Remote) state 706;

From Single Talk (Local) state 704 to either No Talking state 702,Double Talk (Local Wins) state 708, or Double Talk (Remote Wins) state710;

From Single Talk (Remote) state 706 to either No Talking state 702,Double Talk (Local Wins) state 708, or Double Talk (Remote Wins) state710;

From Double Talk (Local Wins) state 708 to only Single Talk (Local)state 704; and

From Double Talk (Remote Wins) state 710 to only Single Talk (Remote)state 706.

Thus, for example, if the state for the previous cycle was the NoTalking state 702, then, during the current cycle, the processing fordiscrete state machine 700 needs only determine whether the conditionsfor either the Single Talk (Local) state 704 or the Single Talk (Remote)state 706 are met. If one of those two sets of conditions are met, thenthe corresponding state transition will occur. Those skilled in the artwill understand that this discrete state machine processing differs fromthe linear state machine processing of FIG. 5 in which the conditionsfor all five modes may need to be checked in a given cycle.

In one embodiment of the present invention, the transmit and receiveattenuation levels for an audio system using discrete state machine 700of FIG. 7 are determined in the same manner as the transmit and receiveattenuation levels that were described earlier for the audio systemusing the state machine processing of FIG. 5. That is, the transmitattenuation level α and the receive attenuation level (1-α) are based onthe state variable i and constant k, where:

for i<-k, α=0

for -k≦i≦k, α=(i+k)/2k

for k<i, α=1

and where i is limited to the range (-j,j). In one embodiment, (j=8) and(k=2).

When a state transition occurs (and only when a state transitionoccurs), the state variable i is incremented (or decremented). Thechange to the state variable i for each of the allowable statetransitions is indicated in FIG. 7 as either a slow step or a fast step.In one embodiment of the present invention, the slow step has amagnitude of 1 and the fast step has a magnitude of l, where l isgreater than k (e.g., l=10).

Those skilled in the art will understand that, in addition to the audioprocessing described above, local node 302 of FIG. 3 may also compressthe local audio signals for transmission to the remote node anddecompress compressed remote audio signals for playback.

It will also be understood that alternative embodiments of the presentinvention may be based on the pre-attenuated local audio signals M ofFIG. 3 rather than on the attenuated local audio signals T.

In one embodiment of the present invention, the network 110 of FIG. 3 isan ISDN network. In this embodiment, the local and remote audio signalsare transmitted between the nodes over the B channels of the ISDNnetwork. To avoid redundant operations between the two nodes and toavoid audio codec distortions to the energy levels, the audio frameenergy (T or R) is embedded within a field of the communication formatthat carries the digitally compressed audio signals for each frame. Thefollowing information is transmitted infrequently over the B channels ofthe ISDN network:

The local background noise threshold τ_(L) sent from the local node tothe remote node; and

The remote background noise threshold τ_(R) sent from the remote node tothe local node.

The infrequent communications are transmitted less frequently than theframe communications (e.g., typically every 5 seconds rather than every20 milliseconds).

In the embodiment shown in FIG. 3, the local node is an open audiosystem while the remote node is a closed audio system. In anotherembodiment of the present invention, both nodes are open audio systems,in which case the remote node performs processing analogous to thatdescribed above for the local node. In this latter embodiment, the localbackground noise threshold τ_(L) is transmitted from the local node tothe remote node in addition to the remote background noise thresholdτ_(R) being transmitted from the remote node to the local node.

In yet another embodiment of the present invention, the remote node doesnot have a histogram generator. In this embodiment, the local nodeperforms the histogram analysis of the remote audio stream received fromthe remote node to generate and update the remote background noisethreshold.

Those skilled in the art will understand that the audio systems of thepresent invention permit two personal computer (PC) based nodesconnected by a digital network to carry on an audio conversationutilizing microphones and speakers without having to resort to othermeasures which limit or eliminate acoustic feedback or coupling fromspeaker to microphone. As such, there is no need for an audio headset orsimilar device to eliminate acoustic coupling. Nor is a commercialspeakerphone attachment (i.e., a separate dedicated circuit device)needed which would perform the audio processing off the PC and would addcost and complexity to the system.

The present invention works with any configuration of one or moremicrophones and one or more speakers and not just with a commercialspeakerphone in which the microphone and speaker are fixed within anenclosure.

It will also be understood that, by transmitting threshold and energylevel information between the nodes, the present invention effectivelydistributes computational overhead among the nodes such that redundantsignal processing (e.g., generation of signal volumes and backgroundnoise thresholds) is eliminated. This also helps make the nodes immuneto any losses or distortions that may result from the compression anddecompression processing.

It will be further understood that various changes in the details,materials, and arrangements of the parts which have been described andillustrated in order to explain the nature of this invention may be madeby those skilled in the art without departing from the principle andscope of the invention as expressed in the following claims.

What is claimed is:
 1. An apparatus for processing audio signals,comprising:(a) a transmit attenuator for attenuating local audio signalsgenerated by a local microphone for transmission to a remote node; (b) areceive attenuator for attenuating remote audio signals received fromthe remote node for playback at a local speaker; and (c) a controller,electrically connected to the transmit attenuator and the receiveattenuator, for controlling a level of transmit attenuation for thetransmit attenuator and a level of receive attenuation for the receiveattenuator, wherein:the controller receives, from the remote node,remote audio information corresponding to the remote audio signals; theremote audio information is distinct from the remote audio signals; theremote audio information has been generated by the remote node based onthe remote audio signals; and the controller uses the remote audioinformation to generate the level of transmit attenuation and the levelof receive attenuation without having to implement local processing ofthe remote audio signals to generate a locally-generated version of theremote audio information.
 2. The apparatus of claim 1, wherein theremote audio information comprises remote energy levels corresponding tothe remote audio signals.
 3. The apparatus of claim 2, wherein theremote audio information further comprises a level of remote backgroundnoise.
 4. The apparatus of claim 3, wherein the level of remotebackground noise is received by the controller from the remote node lessfrequently than the remote energy levels.
 5. The apparatus of claim 1,wherein the remote audio information comprises a level of remotebackground noise.
 6. The apparatus of claim 1, wherein the controllergenerates local audio information corresponding to the local audiosignals for transmission to the remote node.
 7. The apparatus of claim6, wherein the local audio information comprises local energy levelscorresponding to the local audio signals.
 8. The apparatus of claim 6,wherein the local audio information comprises a level of localbackground noise.
 9. The apparatus of claim 1, wherein:the remote audioinformation comprises remote energy levels corresponding to the remoteaudio signals and a level of remote background noise; the level ofremote background noise is received by the controller from the remotenode less frequently than the remote energy levels; the controllergenerates local audio information corresponding to the local audiosignals for transmission to the remote node; the local audio informationcomprises local energy levels corresponding to the local audio signalsand a level of local background noise; and the controller uses the localand remote energy levels and the levels of remote and local backgroundnoise to generate the levels of transmit and receive attenuation.
 10. Acomputer-implemented process for processing audio signals, comprisingthe steps of:(a) receiving, by a controller from a remote node, remoteaudio information corresponding to remote audio signals, where:theremote audio information is distinct from the remote audio signals; andthe remote audio information has been generated by the remote node basedon the remote audio signals; (b) generating, by the controller, a levelof transmit attenuation and a level of receive attenuation using theremote audio information without having to implement local processing ofthe remote audio signals to generate a locally-generated version of theremote audio information; (c) attenuating, with a transmit attenuator atthe level of transmit attenuation, the local audio signals fortransmission to a remote node; and (d) attenuating, with a receiveattenuator at the level of receive attenuation, remote audio signalsreceived from the remote node for playback at a local speaker.
 11. Theprocess of claim 10, wherein step (a) comprises the step of receivingremote energy levels corresponding to the remote audio signals.
 12. Theprocess of claim 11, wherein step (a) further comprises the step ofreceiving a level of remote background noise.
 13. The process of claim12, wherein step (a) comprises the step of receiving the level of remotebackground noise less frequently than the remote energy levels.
 14. Theprocess of claim 10, wherein step (a) comprises the step of receiving alevel of remote background noise.
 15. The process of claim 10, whereinstep (a) further comprises the step of generating local audioinformation corresponding to the local audio signals for transmission tothe remote node.
 16. The process of claim 15, wherein step (a) comprisesthe step of generating local energy levels corresponding to the localaudio signals.
 17. The process of claim 15, wherein step (a) comprisesthe step of generating a level of local background noise.
 18. Theprocess of claim 10, wherein:step (a) comprises the steps of:(1)receiving remote energy levels corresponding to the remote audio signalsand a level of remote background noise, wherein the level of remotebackground noise is received by the controller from the remote node lessfrequently than the remote energy levels; and (2) generating localenergy levels corresponding to the local audio signals and a level oflocal background noise for transmission to the remote node; and step (b)comprises the step of generating the levels of transmit and receiveattenuation using the local and remote energy levels and the levels ofremote and local background noise.